Abstract: Embodiments include methods and systems for improving an acoustic model. Aspects include acquiring a first standard deviation value by calculating standard deviation of a feature from first training data and acquiring a second standard deviation value by calculating standard deviation of a feature from second training data acquired in a different environment from an environment of the first training data. Aspects also include creating a feature adapted to an environment where the first training data is recorded, by multiplying the feature acquired from the second training data by a ratio obtained by dividing the first standard deviation value by the second standard deviation value. Aspects further include reconstructing an acoustic model constructed using training data acquired in the same environment as the environment of the first training data using the feature adapted to the environment where the first training data is recorded.

Abstract: Apparatus and methods for audio classifying and processing are disclosed. In one embodiment, an audio processing apparatus includes an audio classifier for classifying an audio signal into at least one audio type in real time; an audio improving device for improving experience of audience; and an adjusting unit for adjusting at least one parameter of the audio improving device in a continuous manner based on the confidence value of the at least one audio type.

Abstract: In some embodiments, a pitch filter for filtering a preliminary audio signal generated from an audio bitstream is disclosed. The pitch filter has an operating mode selected from one of either: (i) an active mode where the preliminary audio signal is filtered using filtering information to obtain a filtered audio signal, and (ii) an inactive mode where the pitch filter is disabled. The preliminary audio signal is generated in an audio encoder or audio decoder having a coding mode selected from at least two distinct coding modes, and the pitch filter is capable of being selectively operated in either the active mode or the inactive mode while operating in the coding mode based on control information.

Abstract: Noise and distortion reduction in a signal processed through analog circuitry includes providing noise reduction circuitry to reduce signal noise generated by at least one analog circuit element. The noise reduction circuitry is adaptively configured to adjust a rate to apply noise reduction to the signal without introducing unwanted distortion. Distortion reduction circuitry is adaptively configured to adjust a rate to apply distortion reduction to the signal without introducing unwanted noise. The signal is processed through the analog circuitry using the adaptively configured noise reduction circuitry and adaptively configured distortion reduction circuitry to reduce both noise and distortion in the signal.

Abstract: Techniques for indicating to a voice-controlled device that a user is going to provide a voice command to the device. In response to receiving such an indication, the device may prepare to process an audio signal based on sound captured by a microphone of the device for the purpose of identifying the voice command from the audio signal. For instance, a user may utilize a signaling device that includes a button that, when actuated, sends a signal that is received by the voice-controlled device. In response to receiving the signal, a microphone of the voice-controlled device may capture sound that is proximate to the voice-controlled device and may create an audio signal based on the sound. The voice-controlled device may then analyze the audio signal for a voice command of the user or may provide the audio signal to a remote service for identifying the command.

Abstract: A method of environmental noise compensation a speech audio signal is provided that includes estimating a fast audio energy level and a slow audio energy level in an audio environment, wherein the speech audio signal is not part of the audio environment, and applying a gain to the speech audio signal to generate an environment compensated speech audio signal, wherein the gain is updated based on the estimated slow audio energy level when the estimated fast audio energy level is not indicative of an audio event in the audio environment and the estimated gain is not updated when the estimated fast audio energy level is indicative an audio event in the audio environment.

Abstract: A voice enhancement method is disclosed. The method of the present invention is adapted for a distributed system. In the present invention, a plurality of picking devices are disposed in a space for picking voice signal. After determining the positions of the picking devices, an enhancement operation is performed on the waveform signals from the picking devices to generate an enhanced voice signal.

Abstract: Embodiments of the invention concern a method for mitigating interference between near field communication and an audio stream in a mobile user equipment, wherein the method includes a step of replacing at least part of the interfered audio stream by a sound representative of an interfering near field communication transaction.

Abstract: Devices, systems and methods are disclosed for reducing noise in input data by performing a hysteresis operation followed by a lateral excitation smoothing operation. For example, an audio signal may be represented as a sequence of feature vectors. A row of the sequence of feature vectors may, for example, be associated with the same harmonic of the audio signal at different points in time. To determine portions of the row that correspond to the harmonic being present, the system may compare an amplitude to a low threshold and a high threshold and select a series of data points that are above the low threshold and include at least one data point above the high threshold. The system may iteratively perform a spreading technique, spreading a center value of a center data point in a kernel to neighboring data points in the kernel, to further reduce noise.

Abstract: Embodiments disclosed herein provide a wireless controller which shows a voice, motion, or an image complying with or not complying with a user's command and controls an external device in accordance with the user's command. According to an embodiment, a wireless controller includes a main body provided in a shape of a flowerpot, and includes a voice recognition unit, a control unit generating a signal for controlling an object to be controlled, which is designated by a voice recognized in the voice recognition unit, in accordance with the voice, and a communication unit outputting the control signal generated in the control unit to the object to be controlled; and an indicator provided at the main body in a shape of at least one of a stem, a leaf, a flower, and a tree, and showing a motion corresponding to the voice recognized in the voice recognition unit.

Abstract: A method and a media processing system (MPS) for automatic gain control are provided. The MPS detects a portion of an input signal corresponding to a measurement window associated with a predetermined time interval and determines a peak value of the input signal during a first predetermined time segment within the measurement window. The MPS determines a number of occurrences of the peak value during a second predetermined time segment within the measurement window and compares the determined number of occurrences of the peak value with a predefined range of occurrences for identifying a speech portion or a non-speech portion of the input signal corresponding to the measurement window. The MPS adjusts a gain corresponding to the first predetermined time segment based on the peak value, upon identifying the speech portion. The MPS iteratively performs the method by shifting the measurement window in steps of the first predetermined time segment.

Abstract: The invention provides a layered audio coding format with a monophonic layer and at least one sound field layer. A plurality of audio signals is decomposed, in accordance with decomposition parameters controlling the quantitative properties of an orthogonal energy-compacting transform, into rotated audio signals. Further, a time-variable gain profile specifying constructively how the rotated audio signals may be processed to attenuate undesired audio content is derived. The monophonic layer may comprise one of the rotated signals and the gain profile. The sound field layer may comprise the rotated signals and the decomposition parameters. In one embodiment, the gain profile comprises a cleaning gain profile with the main purpose of eliminating non-speech components and/or noise. The gain profile may also comprise mutually independent broadband gains.

Abstract: A method for performing a surround audio compatibility assessment on a plurality of original surround channels is described herein. A surround audio compatibility assessment system accepting original surround signals is also described herein.

Abstract: Voice trigger. In accordance with a first method embodiment, a long term average audio energy is determined based on a one-bit pulse-density modulation bit stream. A short term average audio energy is determined based on the one-bit pulse-density modulation bit stream. The long term average audio energy is compared to the short term average audio energy. Responsive to the comparing, a voice trigger signal is generated if the short term average audio energy is greater than the long term average audio energy. Determining the long term average audio energy may be performed independent of any decimation of the bit stream.

Abstract: The invention provides an audio signal processing device capable of improving sound quality by causing a voice switch to operate appropriately. Delay-subtraction processing is performed on an input signal to form a first and second directional signal with nulls in a first and second specific direction, respectively, and a coherence is obtained using the two directional signals. The coherence is then compared to a determination threshold value to determine whether the input audio signal is a target-sound segment arriving from a target-direction, or a non-target-sound segment other than the target-sound segment. A gain is set according to the determination result, and any non-target-sound is attenuated by multiplying the input signal by the gain. The determination threshold value is controlled based on an average value of coherence in interfering-sound segments.

Abstract: Providing stimulation signals for an implanted auditory prosthesis including receiving first and second sound signals at first and second sound input devices, each of the first and second signals having a signal-to-noise ratio; determining a signal parameter related to said signal-to-noise ratio of each of the first and second signals; selecting one of the first and second signals which has the greater signal-to-noise ratio; and generating stimulation signals for the implanted auditory prosthesis based on said selected sound signal.

Abstract: A system and method for suppressing noise in one or more of at least first and second channels include obtaining a magnitude difference of signals in the first and second channels, obtaining a magnitude sum of signals in the first and second channels, obtaining a ratio of the magnitude difference to the magnitude sum, generating an attenuation value based on the ratio, selecting an attenuator based on the magnitude difference, and attenuating a signal in a channel by the attenuation value using the selected attenuator.

Abstract: In some embodiments, a method for adaptive control of gain applied to an audio signal, including steps of analyzing segments of the signal to identify audio objects (e.g., voices of participants in a voice conference); storing information regarding each distinct identified object; using at least some of the information to determine at least one of a target gain, or a gain change rate for reaching a target gain, for each identified object; and applying gain to segments of the signal indicative of an identified object such that the gain changes (typically, at the gain change rate for the object) from an initial gain to the target gain for the object. The information stored may include a scene description. Aspects of the invention include a system configured (e.g., programmed) to perform any embodiment of the inventive method.

Abstract: In encoding, index information indicating a group of coefficients that minimizes the sum of the error between the value of each sample and the value is obtained by multiplying the quantized value of each of a plurality of samples by a coefficient corresponding to the position of the sample. The coefficient is selected from a plurality of groups of predetermined coefficients corresponding to the positions of the samples. In decoding, a plurality of values corresponding to an input vector quantization index are obtained as decoded values corresponding to a plurality of sample positions. With the use of a group of predetermined coefficients corresponding to the plurality of sample positions and indicated by input index information, the values obtained by multiplying the decoded values and the coefficients, corresponding to the sample positions are output.

Abstract: A voice quality enhancement (VQE) detector for a network element receiving an audio signal from a previous network element of a network, wherein said voice quality enhancement detector is adapted: to perform a voice quality enhancement detection based on the received audio signal, wherein said voice quality enhancement detection comprises detecting that at least one voice quality enhancement function, VQEE comprising of a noise cancellation or an echo cancellation function using a Gaussian Mixture Model was applied to the received audio signal by at least one previous network element of the network; and to control a voice quality enhancement processing of the received audio signal depending on the detection result.

Abstract: An envelope tracking transceiver dynamically adjusts envelope tracking parameters to achieve the desired tradeoff between noise performance and power efficiency. When higher levels of noise are acceptable, the envelope tracking transceiver dynamically adjusts transmitter parameters to achieve better power efficiency while sacrificing noise performance. When lower levels of noise are desired, the envelope tracking transceiver dynamically adjusts parameters to achieve better noise performance while sacrificing efficiency.

Abstract: A method of operating a device for treating sleep disordered breathing (SDB), wherein the device provides continuous positive airway pressure during sleep, includes applying a treatment pressure to a patient, monitoring the patient for speech output, generating a signal in response to detected speech of the patient, and, in response to the signal, reducing the treatment pressure applied to the patient.

Abstract: A method on a mobile device for providing an audio output to a user at a determined loudness level is described. A text input is received. A loudness level of an audio output, corresponding to the text input, is determined based on a volume setting of the mobile device and a non-linear adjustment of the volume setting. The audio output is provided to an audio output component at the determined loudness level.

Abstract: A speech processing apparatus and method. The speech processing apparatus includes a microphone to receive a speech signal, an analog/digital converter to convert the speech signal generated by the microphone into a digital speech signal, and an automatic gain controller to calculate an average value of the magnitude of the digital speech signal generated by the analog/digital converter in a plurality of frames, to determine in which region of a speech signal band the average value is located, the speech signal band being divided into a plurality of regions according to the strength of speech, and to adjust gain according to a location of the average value on the speech signal band so that the strength of speech has a level of an optimal region capable of processing the speech signal. Accordingly, speech recognition may be maximized without being constrained by the distance of a speech source.

Abstract: Volume extension includes limiting the magnitude of Fast Fourier Transform (FFT) frequency bins which allows increases to the perceived level of audio content without causing distortion. A soft limit and smoothing is applied to each FFT bin is to prevent or reduce distortion while maximizing output volume. Frequency resolution is significantly improved compared to volume extension methods utilizing filterbanks and hard limiting, and distortion is reduced because no hard limiting occurs.

Abstract: An apparatus and method for encoding and decoding a signal for high frequency bandwidth extension are provided. An encoding apparatus may down-sample a time domain input signal, may core-encode the down-sampled time domain input signal, may transform the core-encoded time domain input signal to a frequency domain input signal, and may perform bandwidth extension encoding using a basic signal of the frequency domain input signal.

Abstract: The present invention relates to an encoding device and method, and a decoding device and method, and a program which enable music signals to be played with higher sound quality by expanding a frequency band. A band pass filter divides an input signal into multiple subband signals, a feature amount calculating circuit calculates feature amount using at least any one of the divided multiple subband signals and the input signal, a high-frequency subband power estimating circuit calculates an estimated value of high-frequency subband power based on the calculated feature amount, and a high-frequency signal generating circuit generates a high-frequency signal component based on the multiple subband signals divided by the band pass filter and the estimated value of the high-frequency subband power calculated by the high-frequency subband power estimating circuit.

Abstract: An audio-based system may perform dynamic level adjustment by detecting voice activity in an input signal and evaluating voice levels during periods of voice activity. The current voice level is compared to a plurality of thresholds to determine a corresponding gain strategy, and the input signal is scaled in accordance with this gain strategy. Further adjustment to the signal is performed to reduce output clipping that might otherwise be produced.

Abstract: Efficient Context Classification and Gated Loudness Estimation The present document relates to methods and systems for encoding an audio signal. The method comprises determining a spectral representation of the audio signal. The determining a spectral representation step may comprise determining modified discrete cosine transform, MDCT, coefficients, or a Quadrature Mirror Filter, QMF, filter bank representation of the audio signal. The method further comprises encoding the audio signal using the determined spectral representation; and classifying parts of the audio signal to be speech or non-speech based on the determined spectral representation. Finally, a loudness measure for the audio signal based on the speech parts is determined.

Abstract: A speech enhancement system controls the gain of an excitation signal to prevent uncontrolled gain adjustments. The system includes a first device that converts sound waves into operational signals. An ambient noise estimator is linked to the first device and an echo canceller. The ambient noise estimator estimates how loud a background noise would be near the first device before or after an echo cancellation. The system then compares the ambient noise estimate to a current ambient noise estimate near the first device to control a gain of an excitation signal.

Abstract: An adaptive equalization system that adjusts the spectral shape of a speech signal based on an intelligibility measurement of the speech signal may improve the intelligibility of the output speech signal. Such an adaptive equalization system may include a speech intelligibility measurement module, a spectral shape adjustment module, and an adaptive equalization module. The speech intelligibility measurement module is configured to calculate a speech intelligibility measurement of a speech signal. The spectral shape adjustment module is configured to generate a weighted long-term speech curve based on a first predetermined long-term average speech curve, a second predetermined long-term average speech curve, and the speech intelligibility measurement. The adaptive equalization module is configured to adapt equalization coefficients for the speech signal based on the weighted long-term speech curve.

Abstract: A system including a first power measuring module, a gain control module, and a signal processing module. The first power measuring module is configured to generate a first power measurement of a signal received by a receiver of a wireless device based on a plurality of reference signals received in the signal by the receiver of the wireless device. Each of the plurality of reference signals is transmitted at a predetermined power. The first power measurement is generated in frequency domain. The first power measurement is generated during a first frame of the signal received by the receiver of the wireless device. The gain control module is configured to adjust a gain of the receiver of the wireless device based on the first power measurement. The signal processing module is configured to process a second frame in the signal subsequent to the first frame at the adjusted gain.

Abstract: A digital log gain to digital linear gain multiplier is disclosed. The digital log gain to digital linear gain multiplier includes a log gain splitter adapted to split a log gain input into an integer log part and a remainder log part. A log scale-to-linear scale converter is adapted to output a linear gain value in response to the integer log part and the remainder log part. A gain multiply circuit is adapted to multiply a digital signal by the linear gain value to output a gain-enhanced digital signal.

Abstract: A voice processing apparatus includes a voice signal acquiring unit that acquires a voice signal converted to plural frequency bands from an input signal having a narrowed band; an expanding unit that generates based on a narrowband component of the voice signal acquired by the voice signal acquiring unit, an expansion band component expanding the band of the voice signal; a correcting unit that corrects the power of the expansion band component by a correction amount determined based on a noise component included in the voice signal acquired by the voice signal acquiring unit; and an output unit that outputs the voice signal of which the band has been expanded based on the expansion band component corrected by the correcting unit and based on the narrowband component of the voice signal acquired by the voice signal acquiring unit.

Abstract: Method and apparatus for audio intelligibility enhancement and computing apparatus are provided. The method includes the following steps. Environment noise is detected by performing voice activity detection according to a detected audio signal from at least a microphone of a computing device. Noise information is obtained according to the detected environment noise and a first audio signal. A second audio signal is outputted by boosting the first audio signal under an adjustable headroom by the computing device according to the noise information and the first audio signal.

Abstract: The application relates to a method of providing a speech intelligibility predictor value for estimating an average listener's ability to understand of a target speech signal when said target speech signal is subject to a processing algorithm and/or is received in a noisy environment. The application further relates to a method of improving a listener's understanding of a target speech signal in a noisy environment and to corresponding device units. The object of the present application is to provide an alternative objective intelligibility measure, e.g. a measure that is suitable for use in a time-frequency environment. The invention may e.g. be used in audio processing systems, e.g. listening systems, e.g. hearing aid systems.

Abstract: An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.

Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between portions of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: A method for processing multichannel acoustic signals which processes input signals of a plurality of channels including the voices of a plurality of speaking persons. The method is characterized by detecting the voice section of each speaking person or each channel, detecting overlapped sections wherein the detected voice sections are common between channels, determining a channel to be subjected to crosstalk removal and the section thereof by use of at least voice sections not including the detected overlapped sections, and removing crosstalk in the sections of the channel to be subjected to the crosstalk removal.

Abstract: A howling canceller which suppresses occurrence of howling even when an open loop gain exceeds “1” in the whole reproduction band. In the howling canceller, an adaptive filter (107) operates a digital received voice signal with a tap coefficient to generate a pseudo echo; a subtractor (108) subtracts the pseudo echo from a digital transmitted voice signal to generate a residual signal; and an amplitude limiting circuit (110) limits the absolute value of the amplitude of the digital received voice signal to be equal to or smaller than a predetermined threshold which ensures that all of a D/A converter (101), a power amplifier (102), a speaker (103), a microphone (104), a microphone amplifier (105), and an A/D converter (106) operate in a linear operation area, and outputs the amplitude-limited digital received voice signal to the D/A converter (101) and the adaptive filter (107).

Abstract: Provided are a method, apparatus, and medium for encoding/decoding a high frequency band signal by using a low frequency band signal corresponding to an audio signal or a speech signal. Accordingly, since the high frequency band signal is encoded and decoded by using the low frequency band signal, encoding and decoding can be carried out with a small data size while avoiding deterioration of sound quality.

Abstract: An electronic device for coding a transient frame is described. The electronic device includes a processor and executable instructions stored in memory that is in electronic communication with the processor. The electronic device obtains a current transient frame. The electronic device also obtains a residual signal based on the current transient frame. Additionally, the electronic device determines a set of peak locations based on the residual signal. The electronic device further determines whether to use a first coding mode or a second coding mode for coding the current transient frame based on at least the set of peak locations. The electronic device also synthesizes an excitation based on the first coding mode if the first coding mode is determined. The electronic device also synthesizes an excitation based on the second coding mode if the second coding mode is determined.

Abstract: Systems and methods for detecting features in spoken speech and processing speech sounds based on the features are provided. One or more features may be identified in a speech sound. The speech sound may be modified to enhance or reduce the degree to which the feature affects the sound ultimately heard by a listener. Systems and methods according to embodiments of the invention may allow for automatic speech recognition devices that enhance detection and recognition of spoken sounds, such as by a user of a hearing aid or other device.

Abstract: Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage.

Abstract: Disclosed are an encoding device and a decoding device which suppress the occurrence of pre-echo artifacts and post-echo artifacts caused by a high layer having a low temporal resolution, and which implement high subjective quality encoding and decoding. An encoding device (100) carries out scalable coding comprising a low layer, and a high layer having a lower temporal resolution than that of the low layer. A start point detection unit (or end point detection unit) (150) determines the start point (or end point) of sections of the decoded low layer signal which have audio, and when the start point (or end point) is determined, a second layer encoding unit (160) selects a bandwidth to be excluded from encoding on the basis of the spectral energy from the decoded first layer signal, excludes the selected bandwidth, and encodes an error signal.

Abstract: A method and apparatus for utterance verification are provided for verifying a recognized vocabulary output from speech recognition. The apparatus for utterance verification includes a reference score accumulator, a verification score generator and a decision device. A log-likelihood score obtained from speech recognition is processed by taking a logarithm of the value of the probability of one of feature vectors of an input speech conditioned on one of states of each model vocabulary. A verification score is generated based on the processed result. The verification score is compared with a predetermined threshold value so as to reject or accept the recognized vocabulary.

Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between portions of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: Systems and methods to automatically equalize coloration in speech recordings is provided. In example embodiments, a reference spectral shape based on a reference signal is determined. An estimated spectral shape for an input signal is derived. Using the estimated spectral shape and the reference spectral shape a comparison is performed to determine gain settings. The gain settings comprise a gain value for each filter of a filter system. Using gain values associated with the gain setting, automatic equalization is performed on the input signal.

Abstract: For a media clip that includes audio content, a novel method for performing dynamic range compression of the audio content is presented. The method performs an analysis of the audio content. Based on the analysis of the audio content, the method generates a setting for an audio compressor that compresses the dynamic range of the audio content. The generated setting includes a set of audio compression parameters that include a noise gating threshold parameter (“noise gate”), a dynamic range compression threshold parameter (“threshold”), and a dynamic range compression ratio parameter (“ratio”).

Abstract: An acoustic shock protection device includes a prediction gain estimator and an audio compressor. The prediction gain estimator is configured to analyze a plurality of linear prediction coefficients of an audio signal and determine a category of the audio signal. The audio compressor is coupled to the prediction gain estimator, and the audio compressor is configured to adjust a signal level of the audio signal according to the category of the audio signal.